Paging Groups & Intercom Module Documentation

Module Overview (Technical)
Module Overview (Commercial/Business)
Module Overview (End User/Administrator)
Configuration Fields Reference
Call Flow / Logic Explanation
Common Scenarios & Examples
Limitations & Important Notes
Troubleshooting Tips
Glossary
Suggested UI Improvements

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1. Module Overview (Technical)

What Are Paging Groups?

Paging Groups provide one-way or two-way audio broadcasting to multiple endpoints simultaneously. This includes:

Paging: One-way announcements to groups of phones
Intercom: Two-way communication (duplex mode)
Multicast: Network-level audio broadcasting to compatible devices

The Four Modes

Mode	Description	Use Case
Default	Standard SIP paging to selected extensions	Office announcements
Duplex	Two-way intercom communication	Warehouse communication
Multicast	RTP multicast to IP speakers	Factory floor PA
Hybrid	Both SIP extensions and multicast	Mixed environments

Architecture

Plaintext

┌─────────────────────────────────────────────────────────────────┐
│                    Paging System Architecture                   │
├─────────────────────────────────────────────────────────────────┤
│                                                                 │
│  User dials: *80 (Paging Group Code)                            │
│       │                                                         │
│       ▼                                                         │
│  ┌──────────────────────────────────────────────────────────┐   │
│  │              Paging Group Lookup                         │   │
│  │  SELECT * FROM paging_groups WHERE code = '*80'          │   │
│  │  → Returns mode, members, settings                       │   │
│  └──────────────────────────────────────────────────────────┘   │
│       │                                                         │
│       ▼                                                         │
│  ┌──────────────────────────────────────────────────────────┐   │
│  │              Based on Mode                               │   │
│  │                                                          │   │
│  │  DEFAULT ──────────────────────────────────────────────  │   │
│  │    │                                                     │   │
│  │    ├─ Create conference room                             │   │
│  │    ├─ Set auto-answer headers                            │   │
│  │    ├─ Call each member extension                         │   │
│  │    └─ Bridge caller to conference                        │   │
│  │                                                          │   │
│  │  DUPLEX (Intercom) ────────────────────────────────────  │   │
│  │    │                                                     │   │
│  │    ├─ Same as default                                    │   │
│  │    └─ Enable two-way audio for all participants          │   │
│  │                                                          │   │
│  │  MULTICAST ────────────────────────────────────────────  │   │
│  │    │                                                     │   │
│  │    ├─ Answer caller                                      │   │
│  │    └─ Stream audio to multicast address (239.x.x.x)      │   │
│  │                                                          │   │
│  │  HYBRID ───────────────────────────────────────────────  │   │
│  │    │                                                     │   │
│  │    ├─ Call SIP extensions (like default)                 │   │
│  │    └─ Simultaneously stream to multicast                 │   │
│  │                                                          │   │
│  └──────────────────────────────────────────────────────────┘   │
│                                                                 │
└─────────────────────────────────────────────────────────────────┘

Key Tables

Table	Purpose
`public.paging_groups`	Group definitions (code, mode, settings)
`public.paging_group_members`	Extension memberships
`public.conference_profiles`	Audio profiles for paging conferences

Key Files

File	Description
`admin/apps/web/src/modules/pagingGroups/`	Frontend module
`admin/apps/web/src/i18n/en/pagingGroups.json`	English translations
`telephony/freeswitch/lua/main/paging/`	Lua paging handlers

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2. Module Overview (Commercial/Business)

Business Value

Paging Groups enable instant mass communication:

External Systems	Built-In Paging
Separate PA system	Integrated with PBX
Additional hardware	Uses existing phones
Manual intercom setup	Configured via UI
Limited scalability	Unlimited groups

Use Cases

Office Announcements

- "John Smith, please come to reception"

- Emergency notifications

- Company-wide messages

Warehouse/Manufacturing

- Foreman announcements

- Safety alerts

- Shift change notifications

Intercom Communication

- Reception to back office

- Security to all stations

- Medical triage coordination

School/Campus

- Classroom announcements

- Emergency lockdown

- Bell system integration

Feature Highlights

Feature	Benefit
Auto-Answer	Phones answer automatically—no user action needed
Mute Participants	One-way paging (listeners can't interrupt)
PIN Protection	Restrict who can initiate pages
Recording	Record all announcements for compliance
Multicast	Broadcast to IP speakers without SIP overhead
Skip Busy	Don't interrupt ongoing calls

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3. Module Overview (End User/Administrator)

What Can You Do?

Create paging groups with assigned extension codes
Add/remove member extensions
Configure one-way (paging) or two-way (intercom) mode
Set up multicast for IP speakers
Protect pages with PIN codes
Enable recording of announcements

User Workflow

Plaintext

┌─────────────────────────────────────────────────────────────────┐
│                 Creating a Paging Group                         │
├─────────────────────────────────────────────────────────────────┤
│                                                                 │
│  Step 1: Basic Information                                      │
│  ├─ Code: *80 (dial code)                                      │
│  ├─ Name: "All Office"                                         │
│  └─ Mode: Default (SIP paging)                                 │
│                                                                 │
│  Step 2: Add Members                                            │
│  ├─ Search: "Sales"                                            │
│  ├─ Select: 1001, 1002, 1003, 1004                             │
│  └─ Total: 4 extensions                                        │
│                                                                 │
│  Step 3: Audio Settings                                         │
│  ├─ Auto Answer: ✓ (phones answer automatically)               │
│  ├─ Mute Participants: ✓ (one-way announcement)                │
│  └─ Volume: 0 (default)                                        │
│                                                                 │
│  Step 4: Advanced (Optional)                                    │
│  ├─ PIN Code: 1234 (require PIN to page)                       │
│  ├─ Skip Busy: ✓ (don't interrupt calls)                       │
│  └─ Recording: Off                                             │
│                                                                 │
│  Step 5: Save → Users can now dial *80 to page                 │
│                                                                 │
└─────────────────────────────────────────────────────────────────┘

Quick Tips

TIP

Auto-Answer Required: For paging to work, phones must support auto-answer headers. Most modern SIP phones do.

TIP

Skip Busy: Enable this to avoid interrupting important calls—the busy extension will be skipped.

CAUTION

Multicast Network Config: Multicast requires proper network switch configuration (IGMP snooping).

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4. Configuration Fields Reference

Basic Tab

Field	Description	Example	Required
Code	Dialable extension for paging	`80`	Yes
Name	Group name	`All Office`	Yes
Mode	Paging strategy	Default, Duplex, Multicast, Hybrid	Yes
Description	Internal notes	`Floor 1 announcements`	No
Enabled	Active/inactive	On/Off	Yes

Members

Field	Description	Notes
Members	Extensions to receive pages	Multi-select from available extensions

Audio & Media

Field	Description	Default
Codec	Audio codec	PCMU (G.711)
Volume	Playback volume adjustment	0
Hold Music	Music while waiting	Default MOH
Announcement Path	Pre-page announcement file	None
Caller ID Name	Display name for recipients	Paging ${ext}

Session Behavior

Field	Description	Default
Timeout	Max wait time (seconds)	20
Auto Answer	Force phones to auto-answer	On
Skip Busy	Skip busy extensions	Off
Mute Participants	One-way audio only	On
Intercom Mode	Enable two-way audio	Off
PIN Code	Require PIN to page	None

Recording

Field	Description	Default
Record Paging	Enable recording	Off
Recording Path	Where to save recordings	/recordings/paging

Multicast Configuration

Field	Description	Example
Multicast Address	IP multicast address	239.1.1.1
Port	UDP port for RTP	5000
Mode	Speak (send) or Listen (receive)	Speak
Interface	Network interface	eth0
TTL	Time-to-live (router hops)	1
Codec	Audio codec for stream	PCMU

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5. Call Flow / Logic Explanation

Standard Paging Flow (Default Mode)

Plaintext

┌─────────────────────────────────────────────────────────────────┐
│                  Standard Paging Flow                           │
├─────────────────────────────────────────────────────────────────┤
│                                                                 │
│  1. User dials *80 (paging group code)                          │
│     │                                                           │
│     ▼                                                           │
│  2. Check PIN requirement (if configured)                       │
│     ├─ PIN required → Prompt for PIN                           │
│     │   ├─ Correct → Continue                                  │
│     │   └─ Wrong → "Invalid PIN" + hangup                      │
│     └─ No PIN → Continue                                       │
│     │                                                           │
│     ▼                                                           │
│  3. Play pre-announcement (if configured)                       │
│     │                                                           │
│     ▼                                                           │
│  4. Create dynamic conference for this page                     │
│     │                                                           │
│     ▼                                                           │
│  5. For each member extension:                                  │
│     ├─ Check if busy (if skip_busy enabled)                     │
│     ├─ Set auto-answer SIP headers                              │
│     └─ Bridge to conference (muted if mute_participants)        │
│     │                                                           │
│     ▼                                                           │
│  6. Wait for specified timeout for endpoints to answer          │
│     │                                                           │
│     ▼                                                           │
│  7. Caller speaks → Audio broadcast to all answered phones      │
│     │                                                           │
│     ▼                                                           │
│  8. Caller hangs up → Conference ends, all legs disconnect      │
│                                                                 │
└─────────────────────────────────────────────────────────────────┘

Multicast Flow

Plaintext

┌─────────────────────────────────────────────────────────────────┐
│                    Multicast Paging Flow                        │
├─────────────────────────────────────────────────────────────────┤
│                                                                 │
│  1. User dials *85 (multicast paging group)                     │
│     │                                                           │
│     ▼                                                           │
│  2. Answer caller                                               │
│     │                                                           │
│     ▼                                                           │
│  3. Start RTP stream to multicast address:                      │
│     Address: 239.255.10.10:5000                                 │
│     Codec: PCMU (G.711µ)                                        │
│     │                                                           │
│     ▼                                                           │
│  4. IP speakers (subscribed to 239.255.10.10) receive audio     │
│     │                                                           │
│     ▼                                                           │
│  5. Caller hangs up → Stream stops                              │
│                                                                 │
└─────────────────────────────────────────────────────────────────┘

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6. Common Scenarios & Examples

Scenario 1: Office-Wide Announcement

Setup:

Setting	Value
Code	80
Name	All Office
Mode	Default
Members	1001-1050 (all office extensions)
Auto Answer	✓
Mute Participants	✓
Skip Busy	✓

Result: Dial 80, all available phones auto-answer (muted), announcement plays.
Scenario 2: Warehouse Intercom
Setup:
Setting Value
Code 70
Name Warehouse Intercom
Mode Duplex
Members 2001, 2002, 2003 (warehouse phones)
Intercom Mode ✓
Mute Participants ✗ (allow responses)
Result: Dial 70, all warehouse phones connect for two-way conversation.
Scenario 3: Factory Floor PA System
Setup:
Setting Value
Code 85
Name Factory PA
Mode Multicast
Multicast Address 239.255.10.10
Port 5000
Codec PCMU
Result: Dial 85, audio streams to all IP speakers configured for that multicast group.
Scenario 4: Protected Emergency Announcement
Setup:
Setting Value
Code 911
Name Emergency Broadcast
Mode Hybrid
PIN Code 9911
Members ALL extensions
Multicast (also to PA speakers)
Recording ✓
Result: Only authorized users with PIN can trigger emergency broadcast to all phones and speakers. All broadcasts recorded.
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7. Limitations & Important Notes

Technical Limitations

WARNING
Auto-Answer Support: Not all SIP phones support auto-answer. Older or basic phones may not activate.

WARNING
Multicast Network Requirements: Multicast requires IGMP-enabled switches and proper network configuration.

IMPORTANT
Codec Compatibility: Multicast endpoints must support the configured codec. PCMU (G.711µ) has widest support.

Best Practices

Test Auto-Answer: Verify phones actually auto-answer before relying on paging

Network Planning for Multicast: Work with network team to enable IGMP snooping

PIN Protection: Use PINs for high-impact pages (emergency, company-wide)

Skip Busy: Always enable to avoid interrupting important calls

Keep Groups Focused: Create smaller, targeted groups rather than one huge group

Phone Configuration

For auto-answer to work, phones typically need:

Auto-answer header support enabled

Intercom mode enabled in phone settings

Correct SIP headers: Alert-Info: info=auto-answer or similar

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8. Troubleshooting Tips

Common Issues

Symptom Possible Cause Solution
Phone doesn't answer Auto-answer not supported Check phone model/settings
No audio on page Codec mismatch Verify codec compatibility
Multicast not working Network issue Check IGMP, switch config
Only some phones answer Skip busy + busy phones Expected behavior
"Invalid PIN" Wrong PIN entered Verify correct PIN
Recording not created Path permissions Check directory permissions
Diagnostic SQL
List paging groups:
Sql
SELECT pg.code, pg.name, pg.mode, pg.enabled, COUNT(m.id) as member_count FROM public.paging_groups pg LEFT JOIN public.paging_group_members m ON pg.id = m.paging_group_id WHERE pg.domain_id = [domain_id] GROUP BY pg.id ORDER BY pg.code;

Check members of a group:
Sql
SELECT pgm.extension, e.description FROM public.paging_group_members pgm JOIN public.sip_extensions e ON pgm.extension = e.extension WHERE pgm.paging_group_id = [group_id];

FreeSWITCH Diagnostics
Check active conferences (paging uses conferences):
Bash
fs_cli -x "conference xml_list"

Debug multicast:
Bash
# Check if multicast traffic is being sent tcpdump -i eth0 host 239.255.10.10 and udp port 5000

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9. Glossary

Term Definition
Paging One-way audio broadcast to multiple phones
Intercom Two-way audio communication between phones
Multicast Network-level broadcast using IP multicast (239.x.x.x addresses)
Auto-Answer Phone feature that answers incoming calls automatically
Duplex Two-way audio communication mode
IGMP Internet Group Management Protocol—enables multicast on switches
TTL Time To Live—how many network hops multicast packets can travel
PCMU G.711µ audio codec—widely supported, good quality
RTP Real-time Transport Protocol—carries audio over IP networks
PA System Public Address system—speakers for announcements
Barge-In Ability to interrupt an ongoing announcement
Skip Busy Don't page extensions that are on active calls
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10. Suggested UI Improvements

The i18n tooltips in this module are already very comprehensive.

UX Enhancements

Phone Compatibility Indicator: Show which member phones support auto-answer

Quick Test Button: "Test Page" button to verify group works

Member Count Badge: Show member count on list page

Audio Preview: Play announcement preview before saving

Network Diagram: Visual representation of multicast routes

Active Pages Widget: Dashboard showing currently active pages

Additional Features to Consider

Feature Benefit
Scheduled Pages Automatic announcements at set times
Zone Chaining Page multiple groups in sequence
Priority Override Emergency pages interrupt everything
TTS Integration Text-to-speech for typed announcements
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Lua Script Status ℹ️

Integration Module

Component Status Notes
Paging Groups ℹ️ Configuration module Paging managed natively by FreeSWITCH mod_conference
--- Documentation last updated: January 2026

Setting	Value
Code	70
Name	Warehouse Intercom
Mode	Duplex
Members	2001, 2002, 2003 (warehouse phones)
Intercom Mode	✓
Mute Participants	✗ (allow responses)

Setting	Value
Code	85
Name	Factory PA
Mode	Multicast
Multicast Address	239.255.10.10
Port	5000
Codec	PCMU

Setting	Value
Code	911
Name	Emergency Broadcast
Mode	Hybrid
PIN Code	9911
Members	ALL extensions
Multicast	(also to PA speakers)
Recording	✓

Symptom	Possible Cause	Solution
Phone doesn't answer	Auto-answer not supported	Check phone model/settings
No audio on page	Codec mismatch	Verify codec compatibility
Multicast not working	Network issue	Check IGMP, switch config
Only some phones answer	Skip busy + busy phones	Expected behavior
"Invalid PIN"	Wrong PIN entered	Verify correct PIN
Recording not created	Path permissions	Check directory permissions

Term	Definition
Paging	One-way audio broadcast to multiple phones
Intercom	Two-way audio communication between phones
Multicast	Network-level broadcast using IP multicast (239.x.x.x addresses)
Auto-Answer	Phone feature that answers incoming calls automatically
Duplex	Two-way audio communication mode
IGMP	Internet Group Management Protocol—enables multicast on switches
TTL	Time To Live—how many network hops multicast packets can travel
PCMU	G.711µ audio codec—widely supported, good quality
RTP	Real-time Transport Protocol—carries audio over IP networks
PA System	Public Address system—speakers for announcements
Barge-In	Ability to interrupt an ongoing announcement
Skip Busy	Don't page extensions that are on active calls

Feature	Benefit
Scheduled Pages	Automatic announcements at set times
Zone Chaining	Page multiple groups in sequence
Priority Override	Emergency pages interrupt everything
TTS Integration	Text-to-speech for typed announcements

Paging Groups & Intercom Module Documentation

Table of Contents

1. Module Overview (Technical)

What Are Paging Groups?

The Four Modes

Architecture

Key Tables

Key Files

2. Module Overview (Commercial/Business)

Business Value

Use Cases

Feature Highlights

3. Module Overview (End User/Administrator)

What Can You Do?

User Workflow

Quick Tips

4. Configuration Fields Reference

Basic Tab

Members

Audio & Media

Session Behavior

Recording

Multicast Configuration

5. Call Flow / Logic Explanation

Standard Paging Flow (Default Mode)

Multicast Flow

6. Common Scenarios & Examples

Scenario 1: Office-Wide Announcement

Scenario 2: Warehouse Intercom

Scenario 3: Factory Floor PA System

Scenario 4: Protected Emergency Announcement

7. Limitations & Important Notes

Technical Limitations

Best Practices

Phone Configuration

8. Troubleshooting Tips

Common Issues

Diagnostic SQL

FreeSWITCH Diagnostics

9. Glossary

10. Suggested UI Improvements

The i18n tooltips in this module are already very comprehensive.

UX Enhancements

Additional Features to Consider

Lua Script Status ℹ️

Integration Module